1. Field of the Invention
The present invention relates to a method and system for optimizing VoIP for satellite connection. More particularly, the present invention relates to a method and system for reducing the total bandwidth and number of packets utilized to transmit a voice transmission over a satellite connection.
2. Description of the Related Art
Voice over Internet Protocol (VoIP) telephony is a technology taking the world by storm. VoIP allows people to use their computer's Internet connection as a telephone, resulting in huge savings on both local and long distance calling. VoIP telephony sends voice transmissions over the Internet as a data packet, which is organized and decoded by VoIP software. Most people who receive a VoIP call over their standard telephone line never even realize that they are connected with a VoIP user. With satellite VoIP you are allowed to use the Internet as a telephone no matter where you are. VoIP over satellite works just like the telephone you have in your home today. You pick up the phone, dial the number and it connects you to whom you're calling through your high-speed satellite internet connection. Satellite VoIP telephony is perhaps the last frontier in VoIP development. Satellite Internet is becoming more and more popular among customers living in remote areas where telephone or cable service has not yet become available, such as drilling station, ships, scientific expeditions, and the like.
The use of satellite Internet service has been limited due to its high cost and technological limitations that include the amount of available bandwidth, which can be as small as 20 kbps, and the amount of packets that can be transmitted per second. Techniques for compressing audio transmissions have been developed to reduce the amount of network traffic used for satellite Internet service. One popular digital audio compression algorithm is G.729. However, the G.729 digital audio compression algorithm is not royalty free. The Internet Low Bit-rate Codec (“iLBC”) audio compression algorithm, on the other hand, is royalty free and gaining some momentum.
Typically, for digital voice/audio transmission, an actual audio signal is divided into smaller packets that are transmitted independently. In VoIP telephony, this division of an audio signal into smaller Real-time Transport Protocol (hereinafter “RTP”) packets is called packetization (hereinafter “ptime”). Currently, the common packet size stores 20 milliseconds of a digital voice/audio transmission. However, the use of this ptime generally provides a network bandwidth of 25 kbps and requires the transmission of more packets than is necessary for an audio signal.
One existing solution used to increase bandwidth and reduce the number of packets necessary to transmit an audio signal is a system where the VoIP service provider connects its media gateway located on the service provider's premises to the PSTN network using a Primary Rate Interface (hereinafter “PRI”). However, this solution suffers from the drawback that the service provider has to maintain the PRI, which results in added expense for lines and equipment maintenance. In addition, connectivity over PRIs does not provide the most effective and cost efficient way to terminate and originate calls. For example, in the case of call termination there is added cost. While, in the case of call origination, where customers are provided with inbound phone numbers, a low capacity network is required to be built with extremely wide presence in different areas.
Another solution is based on existing VoIP practice, where VoIP service providers utilize different PSTN carriers to originate and terminate call traffic in the most optimal way. However, this solution suffers from the drawback that termination and origination carriers cannot properly handle VoIP traffic with a ptime that is not 20 ms. Moreover, in the case of the G.729 compression algorithm, the use of a ptime that is not 20 ms is non-compliant with existing high capacity switches produced by companies such as Sonus, Lucent and others. In the case of iLBC it happens not to be supported at all. There is a need to decrease the actual bandwidth used during satellite VoIP. There is also a need to reduce the number of packet necessary to transmit an audio signal.